ALERT GVSIP Is Dead, was: GVSIP Registration Issues. Asterisk 13.23.0 Seems To Fix The Problem.

[wardmundy]

I'm just wondering if some small tweak could be done to the way pjsip registers with google instead of this massive reinstall operation to fix the problem.

Follow the issue here. We're all dead at the moment except for pbxes.com users.

 

[weenus500]

Follow the issue here. We're all dead at the moment except for pbxes.com users.

Thank you,
FWIW: I have an obi200 and this is the only thing working with google voice. This problem just started some time over night. Is that same for every one else?

 

[kdthomas]

Yup.. I wonder what pbxes.com is doing different. Anyone with connections with them? Also following the DSL thread.

 

[wardmundy]

Here's the fix:

sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /etc/asterisk/pjsip_custom.conf
amportal restart
sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /root/gvsip-naf/install-gvsip

 

[lthown]

There was an update posted

"Use voice.telephony.goog instead of obihai.telephony.goog for the sip connection and things seem to work."​

Dunno how to make use of that though

edit: Ward got it!

edit2:

asterisk -rx "pjsip show registrations"
 <Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================
 gvsip1/sip:obihai.sip.google.com                        gvsip1            Registered
 gvsip2/sip:obihai.sip.google.com                        gvsip2            Registered

 

[kdthomas]

sweet jeebus, glad they got that figured out...working now as well

 

[ariban]

Here's the fix:

sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /etc/asterisk/pjsip_custom.conf
amportal restart
sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /root/gvsip-naf/install-gvsip

is this the same as update 730? or should i do this even after update 730?

 

[weenus500]

Here's the fix:

sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /etc/asterisk/pjsip_custom.conf
amportal restart
sed -i 's|obihai.telephony.goog|voice.telephony.goog|' /root/gvsip-naf/install-gvsip

Yep. That's the ticket. Thank you Mr. Ward Mundy.
I am less daring than so my way of fixing it was to edit those two files with vim like so:
:%s/obihai.telephony.goog/voice.telephony.goog/gc
I wanted to show some honest appreciation so I just made a payment to your paypal account. Please keep up the good work.

 

[sortons]

It's all back up and running after applying the fix in post #64. Thank you!

 

[weenus500]

After a number of weeks of problem free service today I woke up to this unhappy reality: "gvsip1/sipbihai.sip.google.com gvsip1 Rejected" on my two raspi servers. I hope that following the procedure for the upgrade to 13.23.1 will get the google voice trunk working again.[/QUO

I'm just wondering if some small tweak could be done to the way pjsip registers with google instead of this massive reinstall operation to fix the problem.

I had problem with step 7. It didn't run at all so being a smart chimp I looked for where it might be on the box and I found these two.
patch -pl < /root/gvsip-naf/gvsip-naf.patch
patch -pl < /usr/src/gvsip-naf.patch

They both came back with this message.

patch: **** strip count l is not a number
I skipped this step 7. and end result is that gvsip wouldn't work.
Maybe some kind soul could help me out with this.

on incrediblepbx asterisk 13.22
unfortunately, I ran the above procedure on my secondary box and it messed it up totally. Now asterisk -rx "pjsip show registration" returns "object not found". Is there anyway to fix this without reinstalling from scratch?

 

[ericlee1]

besides that issue, lately I've been noticing that I don't get a ring while dialing out. just a silence and then it connects, anyone else experience this.. is there a fix for that?

 

[ajonate]

besides that issue, lately I've been noticing that I don't get a ring while dialing out. just a silence and then it connects, anyone else experience this.. is there a fix for that?

What you describe doesn't sound like an asterisk or freepbx issue. It sounds more like an issue with your sip phone, ata device, or softphone. There's usually a setting something like 'generate ringing on outgoing calls."

 

[Vladimir Mikhelson]

I had problem with step 7. It didn't run at all so being a smart chimp I looked for where it might be on the box and I found these two.
patch -pl < /root/gvsip-naf/gvsip-naf.patch
patch -pl < /usr/src/gvsip-naf.patch

They both came back with this message.

patch: **** strip count l is not a number
I skipped this step 7. and end result is that gvsip wouldn't work.
Maybe some kind soul could help me out with this.

Weenus500,

Please make sure the command is spelled “patch -p1....” not “-pl” I used the patch in /usr/src

-Vladimir

 

[Vladimir Mikhelson]

besides that issue, lately I've been noticing that I don't get a ring while dialing out. just a silence and then it connects, anyone else experience this.. is there a fix for that?

Ericlee1,

Unfortunately it is an old and known issue. It happens on certain called numbers, not all. Naf is still working on it.

-Vladimir

 

[Vladimir Mikhelson]

Ok, update!!

I upgraded, i now have 13.23.1 asterisk version
it worked for about a day, and then it disconnected, i ran ./show-trunks and it shows all gvsip as registered (i ran your script too just in case but it came back saying its all registered and nothing to do)
only solution again was to type amportal restart

Ariban,

As I hear you the problem was not fixed by 13.23.1. If that is correct then you have some other issue.

Do you see anything of inetrest in the Asterisk console at the time you experience the issue? How does it compare to what you see after the restart while everything is still working fine?

-Vladimir

 

[shetu]

Today my gv account not register (Aterisk 13.23.1, Scientific linux )

[2018-10-13 08:08:24] WARNING[26667]: res_pjsip_outbound_registration.c:987 schedule_retry: No response received from 'sipbihai.sip.google.com' on registration attempt to 'sip:[email protected]', retrying in '60'

 

[roarsys]

Weenus500,

Please make sure the command is spelled “patch -p1....” not “-pl” I used the patch in /usr/src

-Vladimir

Running on raspi 3b+

I am having the same issue, I did patch it properly using /usr/src file as well as the one included in the gvsip-naf directory. Actually updating Asterisk broke a few things, I think I fixed most of it.

=========================================================================
Connected to Asterisk 13.23.1 currently running on incrediblepbx (pid = 27688)
incrediblepbx*CLI> pjsip show registrations
No objects found.

I just realized it quoted the reply not the issue. Sorry coffee consumption had not begun yet at this time.

 

[ajonate]

Please make sure the command is spelled “patch -p1....” not “-pl” I used the patch in /usr/src

At the risk of stating the obvious, Linux command line recipes are precise enough that followers really need to copy and paste commands, not try to retype them.

 

[Vladimir Mikhelson]

At the risk of stating the obvious, Linux command line recipes are precise enough that followers really need to copy and paste commands, not try to retype them.

Ajonate,

You are 100% correct. I was just replying to the specific post #73.

-Vladimir

 

[ajonate]

Ajonate,

You are 100% correct. I was just replying to the specific post #73.

-Vladimir

That post wasn't really intended for you, although I quoted you.